DS1 has become an industry standard for telecommunication data format. In standard DS1, voice signals are digitalized at 64 kilobits per second (Kb/s), and 24 such 64 Kb/s streams are multiplexed into a 1.544 megabits per second (Mb/s) DS1 signal and transmitted over a T1 trunk. Often a 56 Kb/s or 64 Kb/s digital data signal is substituted for a digitalized voice signal. A waveform encoding technique, commonly referred to as ADPCM (adaptive differential pulse code modulation), compresses voice data to 32 Kb/s and allows a T1 trunk to carry up to 48 voice channels. However, because certain digital data signals of 56 Kb/s or 64 Kb/s (switched-56 and switched-64) are not compressible, the effective compression ratio over standard DS1 is less than 2:1. Standard ADPCM also cannot support voice-band-modem traffic where the baud rate is 9.6 Kb/s or higher.
Recently, other waveform encoding techniques have been developed that encode voice data in 16 Kb/s or 24 Kb/s. In using these techniques, greater than 2:1 compression ratio is achieved, but at the cost of poorer data quality.
Another type of known data compression technique is digital speech interpolation or DSI. The underlying idea of DSI makes use of characteristics of human conversation. More particularly, in a typical telephone conversation between two parties or more, usually only one party is talking at any one time. In other words, the transmission may be unidirectional most of the time. Additionally, human speech pattern usually includes periods of silence, where transmission is not required. Accordingly, DSI enables a reduction of transmission data by utilizing the statistical nature of human speech and conversation. The compression factor for a conventional DSI system is approximately 2:1.
Several disadvantages of pure DSI data compression systems exist. In order to utilize DSI, an accurate determination of silence is required so that actual speech is not deleted. Known "silence" determination schemes introduce significant delays of up to 30 msec in the encoding device and up to 15 msec in the decoding device. Additionally in DSI, the mapping of channels to trunks is variable with the current traffic condition with only the updates of mapping changes being transmitted from the sending device to the receiving device. In the event of a burst error and the like, the updates may be lost and recovery time may be substantial. Lastly, DSI is worthless as a compression scheme for signals that do not have periods of silence.
It is therefore desirable to compress data at a ratio of greater than 2:1 without compromising the quality of data. It is further desirable to carry modem communication data signals of 56 Kb/s and 64 Kb/s. This is accomplished in the present invention by combining waveform encoding technology with DSI.
It is further desirable to determine three key parameters of the data signals being transmitted in order to compress data by a combination of waveform encoding and DSI. These key parameters are: is the data in a channel compressible? Is the channel silent? What is the background noise level?
Accordingly, the present invention provides for apparatus and a method for determining the above three key parameters, for compressing data at a ratio of N:1 without compromising the quality of the data and is directed to overcoming one or more of the problems as set forth above.